<div dir="ltr">It is the way it works. First the phone sends a REGISTER without any password. Asterisk answers with a "Unauthorized" and provide a nonce to be used for the next registration attempt, using it to encrypt the password.<div>
<br></div><div>Leandro</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-05-14 13:12 GMT+02:00 Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div>Hello,</div><div><br></div><div>After a small break from working on this, I got the idea of tcpdumping the correct ports. What I see is REGISTER messages from Kamailio port to Asterisk, which are replied with 401 Unauthorized. Why is this happening? In my sippeers table the secret field has no value (tried both NULL and empty string) and the added field sippasswd has the correct password for the user. </div>
<div><br></div><div>The above might be the cause of my problem, would anyone be able to advice me to get to correct behaviour? Now Kamailio sees the clients as registered, which would be wrong if Asterisk doesn't. </div>
<div><br></div><div>cheers,</div><div>Olli</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-04-24 11:27 GMT+03:00 Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span>:<div>
<div class="h5"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div><div>Hello all,</div><div><br></div><div>I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation.</div>
<div><br></div><div>My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. </div>
<div><br></div><div>In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the user data. Field regseconds has a value and fullcontact has value '<a href="http://sip:660@127.0.0.1:5060" target="_blank">sip:660@127.0.0.1:5060</a>' (kamailio ip:port as they are on the same machine).</div>
<div><br></div><div>I have a very simple dialplan:</div><div><br></div><div><div>[general]</div><div><br></div><div>[default]</div><div>exten => _XXX,1,NoOp(general : Dialed ${EXTEN})</div><div> same => n,Dial(SIP/${EXTEN},3600,rt)</div>
<div> same => n,Hangup</div></div><div><br></div><div><br></div><div>Here's more on my problem and background to it, guys on the Kamailio list helped out but looks like I need to check my Asterisk configuration. </div>
<div><a href="https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html" target="_blank">https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html</a></div><div><br></div><div>My goal is to have all clients in the asterisk database, asterisk (one at this point, several later) handling the calls and Kamailio as proxy. In Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one domain '<a href="http://testers.com" target="_blank">testers.com</a>'. </div>
<div><br></div><div>I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on the same rental virtual server. Clients are in my home network behind nat.</div><div>In MySQL I have database asterisk with table sippeers, where I have clients added like this: </div>
<div>INSERT INTO sippeers (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type) VALUES ('660', '660', 'dynamic', 'password', '660', '<a href="http://testers.com" target="_blank">testers.com</a>','660','friend');</div>
<div><br></div><div>In this message there are some outputs and a sip trace of a register:</div><div><a href="https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html" target="_blank">https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html</a></div>
<div><br></div><div>What I don't know is how to configure sip.conf, so far I've just been making guesses based on online examples and documentation. </div><div>My current sip.conf looks like this:</div><div><br></div>
<div>[general]</div><div>bindport = 5070</div><div>bindaddr = 127.0.0.1</div><div>tcpbindaddr = <a href="http://127.0.0.1:5070" target="_blank">127.0.0.1:5070</a></div><div>tcpenable = no</div><div>limitonpeers = yes</div>
<div>;rtcachefriends = yes</div>
<div>tos_sip=cs3</div><div>tos_audio=ef</div><div>realm = <a href="http://testers.com" target="_blank">testers.com</a></div><div><br></div><div>I've tried defining realm and domain values, but I lack proper understanding of those. Can you guys help me out? Are there any other configurations I need to check? </div>
<div><br></div><div>Respectfully,</div><div>Olli</div></div><div><br></div><div><br></div></div>
</blockquote></div></div></div><br></div>
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